Re: [OpenSIPS-Users] Call from Asterisk to Opensips
Hi Truong first thing you should try to read the asterisk SIP TRUNK and here is the basic example and i think the problem is asterisk not opensips http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/ and make sure to check the debug from asterisk and opensips, i think you will get the clues :D Ha` --- On Thu, 5/5/11, Duong Manh Truong ngoahotanglon...@gmail.com wrote: From: Duong Manh Truong ngoahotanglon...@gmail.com Subject: [OpenSIPS-Users] Call from Asterisk to Opensips To: OpenSIPS users mailling list users@lists.opensips.org Date: Thursday, May 5, 2011, 9:53 PM Hi all, I've created sip trunk on Asterisk and defined asterisk server ip on address table of opensips Then, from extension of Opensips , i can dial out to pstn through Asterisk Now, i want to route PSTN call to the extension but when Asterisk receive the call from PSTN and dial Opensips through the Sip Trunki always got the message in the asterisk's console: Called to-opensips/1001 -- SIP/to-opensips-0745 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) (1001 is the extension of Opensips) Then the call hangs up. Anyone got this problem ? please help me the way to deal with! Thanks so much! -Inline Attachment Follows- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips account: Local calls = Proxy authentication required
Hi Truong please use the pastebin.com to upload your opensips.conf i can check it in details Thanks Ha` --- On Wed, 5/4/11, Duong Manh Truong ngoahotanglon...@gmail.com wrote: From: Duong Manh Truong ngoahotanglon...@gmail.com Subject: Re: [OpenSIPS-Users] Opensips account: Local calls = Proxy authentication required To: OpenSIPS users mailling list users@lists.opensips.org Date: Wednesday, May 4, 2011, 6:07 PM Hi Ha`! Thanks for your help. I read the book and followed steps written there But now i can not figure out why local calls are not successful Please help me in more details! Regards. Hi Truong try the ebook https://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book there are a lot of examples and explain how opensips work hope this help Ha` --- On Wed, 5/4/11, Duong Manh Truong ngoahotanglon...@gmail.com wrote: From: Duong Manh Truong ngoahotanglon...@gmail.com Subject: [OpenSIPS-Users] Opensips account: Local calls = Proxy authentication required To: OpenSIPS users mailling list users@lists.opensips.org Date: Wednesday, May 4, 2011, 4:05 AM Hi all,? After successfully installation of Opensips, my extensions can dial to each other ?(local call)?example: ext 1000 can dial to ext 1001? Then i configure Opensips to go to PSTN through Asterisk (gateway) by using rewritehostport function and address table in opensips database (for Asterisk server IP) Now, i can call from Opensips to PSTN over Asterisk but.for local calls (1000 call 1001),?there're always the message Proxy authentication required in Xlite screen ??? I type opensipsctl online and find that both of 2 exts are online ??? Please tell me how to deal with this matter? ?( I attached my opensips.conf file in this email)? -Inline Attachment Follows- -Inline Attachment Follows- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips account: Local calls = Proxy authentication required
Hi Truong there are some parameters in your opensips.cfg modparam(auth_db, calculate_ha1, yes) modparam(auth_db, password_column, password) modparam(auth_db, db_url, mysql://opensips:opensipsrw@localhost/opensips) modparam(auth_db, load_credentials, ) you use clear text password or encrypted password for your subscribers take time to read and check it again in mysql database Thank you Ha` --- On Thu, 5/5/11, ha do haloha...@yahoo.com wrote: From: ha do haloha...@yahoo.com Subject: Re: [OpenSIPS-Users] Opensips account: Local calls = Proxy authentication required To: OpenSIPS users mailling list users@lists.opensips.org Date: Thursday, May 5, 2011, 2:54 AM Hi Truong please use the pastebin.com to upload your opensips.conf i can check it in details Thanks Ha` --- On Wed, 5/4/11, Duong Manh Truong ngoahotanglon...@gmail.com wrote: From: Duong Manh Truong ngoahotanglon...@gmail.com Subject: Re: [OpenSIPS-Users] Opensips account: Local calls = Proxy authentication required To: OpenSIPS users mailling list users@lists.opensips.org Date: Wednesday, May 4, 2011, 6:07 PM Hi Ha`! Thanks for your help. I read the book and followed steps written there But now i can not figure out why local calls are not successful Please help me in more details! Regards. Hi Truong try the ebook https://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book there are a lot of examples and explain how opensips work hope this help Ha` --- On Wed, 5/4/11, Duong Manh Truong ngoahotanglon...@gmail.com wrote: From: Duong Manh Truong ngoahotanglon...@gmail.com Subject: [OpenSIPS-Users] Opensips account: Local calls = Proxy authentication required To: OpenSIPS users mailling list users@lists.opensips.org Date: Wednesday, May 4, 2011, 4:05 AM Hi all,? After successfully installation of Opensips, my extensions can dial to each other ?(local call)?example: ext 1000 can dial to ext 1001? Then i configure Opensips to go to PSTN through Asterisk (gateway) by using rewritehostport function and address table in opensips database (for Asterisk server IP) Now, i can call from Opensips to PSTN over Asterisk but.for local calls (1000 call 1001),?there're always the message Proxy authentication required in Xlite screen ??? I type opensipsctl online and find that both of 2 exts are online ??? Please tell me how to deal with this matter? ?( I attached my opensips.conf file in this email)? -Inline Attachment Follows- -Inline Attachment Follows- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -Inline Attachment Follows- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips account: Local calls = Proxy authentication required
Hi Truong try the ebook https://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book there are a lot of examples and explain how opensips work hope this help Ha` --- On Wed, 5/4/11, Duong Manh Truong ngoahotanglon...@gmail.com wrote: From: Duong Manh Truong ngoahotanglon...@gmail.com Subject: [OpenSIPS-Users] Opensips account: Local calls = Proxy authentication required To: OpenSIPS users mailling list users@lists.opensips.org Date: Wednesday, May 4, 2011, 4:05 AM Hi all, After successfully installation of Opensips, my extensions can dial to each other (local call) example: ext 1000 can dial to ext 1001 Then i configure Opensips to go to PSTN through Asterisk (gateway) by using rewritehostport function and address table in opensips database (for Asterisk server IP) Now, i can call from Opensips to PSTN over Asterisk but.for local calls (1000 call 1001), there're always the message Proxy authentication required in Xlite screen ??? I type opensipsctl online and find that both of 2 exts are online ??? Please tell me how to deal with this matter? ( I attached my opensips.conf file in this email) -Inline Attachment Follows- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips sometime resend Accouting-request to freeradius
Hi Dani if you look at my opensips.cfg and there is only 1 time for INVITE and 1 time for BYE to do accouting the problem here is sometimes freeradius is not send the Accouting-Response to radiusclient so that is why radiusclient resend the Accouting-request opensips and freeradius run in the same server and freeradius run at interface 127.0.0.1 i use virtual box to test i use the freeradius from Ag-Project in my option, the problem is freeradius Thank you Ha` --- On Thu, 3/24/11, Dani Popa dani.p...@gmail.com wrote: From: Dani Popa dani.p...@gmail.com Subject: Re: [OpenSIPS-Users] Opensips sometime resend Accouting-request to freeradius To: users@lists.opensips.org Date: Thursday, March 24, 2011, 7:11 AM You have a problem with opensips script. You set acct start flag many times and radius try each time to insert in mysql the same start query, but you can not because you have key defined on your table. Dani On 03/23/11 10:06, ha do wrote: Hi list i am test opensips 1.6.4 freeradius from ag-projects cdrtool version 8.0.17 callcontrol version 2.0.8 there is sometimes opensips resend Accounting-Request to freeradius and make freeradius seems to reject message from opensips here is the pastebin link: http://pastebin.com/d1k38n5a - freeradius log http://pastebin.com/ww2Dq3SC - opensips.cfg how to resolve problem Thank you ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips sometime resend Accouting-request to freeradius
Hi list i am test opensips 1.6.4 freeradius from ag-projects cdrtool version 8.0.17 callcontrol version 2.0.8 there is sometimes opensips resend Accounting-Request to freeradius and make freeradius seems to reject message from opensips here is the pastebin link: http://pastebin.com/d1k38n5a - freeradius log http://pastebin.com/ww2Dq3SC - opensips.cfg how to resolve problem Thank you ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips sometime resend Accouting-request to freeradius
Hi i found problem, Thank you Ha` --- On Wed, 3/23/11, ha do haloha...@yahoo.com wrote: From: ha do haloha...@yahoo.com Subject: Opensips sometime resend Accouting-request to freeradius To: OpenSIPS users@lists.opensips.org Date: Wednesday, March 23, 2011, 2:06 AM Hi list i am test opensips 1.6.4 freeradius from ag-projects cdrtool version 8.0.17 callcontrol version 2.0.8 there is sometimes opensips resend Accounting-Request to freeradius and make freeradius seems to reject message from opensips here is the pastebin link: http://pastebin.com/d1k38n5a - freeradius log http://pastebin.com/ww2Dq3SC - opensips.cfg how to resolve problem Thank you ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Munin monitoring plugin
Hi interesting plug_in do you have screen shot of pictures :D --- On Tue, 3/22/11, Henning Holtschneider henn...@loca.net wrote: From: Henning Holtschneider henn...@loca.net Subject: [OpenSIPS-Users] Munin monitoring plugin To: OpenSIPS users mailling list users@lists.opensips.org Date: Tuesday, March 22, 2011, 2:01 AM -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello everybody, I wrote a Munin monitoring plugin which collects data via the 'opensipsctl fifo get_statistics' command the other day. The plugin is available at http://github.com/hehol/OpenSIPS-Munin-Plugin. If you have any questions or suggestions, feel free to contact me or just fork my code at Github and contribute! Cheers, Henning Holtschneider - -- LocaNet oHG - http://www.loca.net Lindemannstrasse 81, D-44137 Dortmund tel +49 231 91596-25, fax +49 231 91596-55 sip 2...@voip.loca.net Registergericht Amtsgericht Dortmund HRA 14208 Geschäftsführer Sven Haufe, Henning Holtschneider -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (Darwin) iEYEARECAAYFAk2IV2EACgkQP9goCV2uudcm3gCfR/37oL7BtYGKdxvvGp9Y2qTS lMMAoO4PV6fO9+WOm9zCNjvdD62QZHdR =+tCa -END PGP SIGNATURE- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need help on connection CDRTool to opensips
Hi Tijmen it works fine right now for callcontrol hangups a call in my last email can i request to make the instruction on config.sample.ini of the OPENSIPS part callcontrol uses the mi_datagram to make it clearer not confusing with mi_fifo and the socket_path is the same socket_name in mi_datagram config of opensips it should be good for someone, likes me i still need help 2 opensips2 is a gateway endpoint-opensips1---opensips2internet-VOIP service provider So basically opensips2 is just a gateway to opensips1? [ha] yes,how to config cdrtool rating calls for opensips2 Thank you Ha - Original Message From: Tijmen de Mes tij...@ag-projects.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Tue, November 30, 2010 3:44:12 PM Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips Hi, On Nov 25, 2010, at 10:06 AM, ha do wrote: and opensips.cfg with fifo config # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0600) modparam(mi_fifo, fifo_group, 0) modparam(mi_fifo, fifo_group, root) modparam(mi_fifo, fifo_user, 0) modparam(mi_fifo, fifo_user, root) and the callcontrol config.ini [CallControl] group = root [OpenSIPS] socket_path = /tmp ;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 terminated i need to increase to 100, right??? how to fix the issue Try using socket_path=/tmp/socket in config.ini. The socket path should match the mi_datagram ,socket_name in the opensips config. Tijmen de Mes AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need help on connection CDRTool to opensips
Hi Tijimen [ha - solved] with help from bogdan_vs: i change the sock_path = /tmp/opensips.sock and use the mi_datagram in opensips.cfg callcontrol hangup calls successfull. So the callcontrol use the mi_datagram, right?? i still need help 2 opensips2 is a gateway endpoint-opensips1---opensips2internet-VOIP service provider So basically opensips2 is just a gateway to opensips1? [ha] yes, Thank you Ha - Original Message From: ha do haloha...@yahoo.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thu, November 25, 2010 4:06:14 PM Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips Hi all can anyone help me on call-control, i already try the mi_datagrambut i get the same error, permission denied Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com started for maximum 60 seconds Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com connected for maximum 60 seconds callcontrol.opensips.UNIXSocketProtocol starting on '/var/run/callcontrol/opensips_01.sock' error: cannot write request to /tmp: Connection refused error: failed to end dialog: Cannot send request to OpenSIPS Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com disconnected by call control after 60 seconds, call price is 0.0998 prepaid account 100 has only 60s to connect the call, and after 60s callcontrol cannot disconnected call the mi_fifo located in /tmp and opensips.cfg with fifo config # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0600) modparam(mi_fifo, fifo_group, 0) modparam(mi_fifo, fifo_group, root) modparam(mi_fifo, fifo_user, 0) modparam(mi_fifo, fifo_user, root) and the callcontrol config.ini [CallControl] group = root [OpenSIPS] socket_path = /tmp ;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 terminated i need to increase to 100, right??? how to fix the issue thank you Ha - Original Message From: ha do haloha...@yahoo.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Wed, November 17, 2010 10:07:16 AM Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips Hi Tijimen sorry for late reply 2 opensips2 is a gateway endpoint-opensips1---opensips2internet-VOIP service provider So basically opensips2 is just a gateway to opensips1? [ha] yes, i just try to see what happen in cdrtool, call from subscriber to subscriber these are the logs Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000 callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39 Gateway=192.168.1.36 To=sip:00842...@192.168.1.39 Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited Type=postpaid callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39 log of call-control Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to sip:00842...@192.168.1.39 is postpaid not limited it always return postpaid You need to specify that a sip account is prepaid if you want it to act like a prepaid account. [ha] it is done and work perfectly except 1 thing Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com started for maximum 60 seconds Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com connected for maximum 60 seconds callcontrol.opensips.UNIXSocketProtocol starting on '/var/run/callcontrol/opensips_01.sock' error: cannot write request to /tmp: Connection refused error: failed to end dialog: Cannot send request to OpenSIPS Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com disconnected by call control after 60 seconds, call price is 0.0998 prepaid account 100 has only 60s to connect the call, and after 60s callcontrol cannot disconnected call the mi_fifo located in /tmp and opensips.cfg with fifo config # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0600) modparam(mi_fifo, fifo_group, 0) modparam(mi_fifo, fifo_group, root) modparam(mi_fifo, fifo_user, 0) modparam(mi_fifo, fifo_user, root) and the callcontrol config.ini [CallControl] group = root [OpenSIPS] socket_path = /tmp ;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 terminated i need to increase to 100, right??? how to fix the issue Thank you Ha` - Original Message From: Tijmen de Mes tij...@ag-projects.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Mon, November 15, 2010 4:11:20 PM Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips Hi, The following steps are performed to rate a CDR: 1. Determination of the billing party
Re: [OpenSIPS-Users] need help on connection CDRTool to opensips
Hi all can anyone help me on call-control, i already try the mi_datagrambut i get the same error, permission denied Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com started for maximum 60 seconds Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com connected for maximum 60 seconds callcontrol.opensips.UNIXSocketProtocol starting on '/var/run/callcontrol/opensips_01.sock' error: cannot write request to /tmp: Connection refused error: failed to end dialog: Cannot send request to OpenSIPS Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com disconnected by call control after 60 seconds, call price is 0.0998 prepaid account 100 has only 60s to connect the call, and after 60s callcontrol cannot disconnected call the mi_fifo located in /tmp and opensips.cfg with fifo config # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0600) modparam(mi_fifo, fifo_group, 0) modparam(mi_fifo, fifo_group, root) modparam(mi_fifo, fifo_user, 0) modparam(mi_fifo, fifo_user, root) and the callcontrol config.ini [CallControl] group = root [OpenSIPS] socket_path = /tmp ;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 terminated i need to increase to 100, right??? how to fix the issue thank you Ha - Original Message From: ha do haloha...@yahoo.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Wed, November 17, 2010 10:07:16 AM Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips Hi Tijimen sorry for late reply 2 opensips2 is a gateway endpoint-opensips1---opensips2internet-VOIP service provider So basically opensips2 is just a gateway to opensips1? [ha] yes, i just try to see what happen in cdrtool, call from subscriber to subscriber these are the logs Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000 callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39 Gateway=192.168.1.36 To=sip:00842...@192.168.1.39 Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited Type=postpaid callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39 log of call-control Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to sip:00842...@192.168.1.39 is postpaid not limited it always return postpaid You need to specify that a sip account is prepaid if you want it to act like a prepaid account. [ha] it is done and work perfectly except 1 thing Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com started for maximum 60 seconds Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com connected for maximum 60 seconds callcontrol.opensips.UNIXSocketProtocol starting on '/var/run/callcontrol/opensips_01.sock' error: cannot write request to /tmp: Connection refused error: failed to end dialog: Cannot send request to OpenSIPS Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com disconnected by call control after 60 seconds, call price is 0.0998 prepaid account 100 has only 60s to connect the call, and after 60s callcontrol cannot disconnected call the mi_fifo located in /tmp and opensips.cfg with fifo config # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0600) modparam(mi_fifo, fifo_group, 0) modparam(mi_fifo, fifo_group, root) modparam(mi_fifo, fifo_user, 0) modparam(mi_fifo, fifo_user, root) and the callcontrol config.ini [CallControl] group = root [OpenSIPS] socket_path = /tmp ;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 terminated i need to increase to 100, right??? how to fix the issue Thank you Ha` - Original Message From: Tijmen de Mes tij...@ag-projects.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Mon, November 15, 2010 4:11:20 PM Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips Hi, The following steps are performed to rate a CDR: 1. Determination of the billing party a. SIP account u...@domain b. SIP domain of the SIP account c. Source IP of the session d. Default (when none of the above matches) if there is no sip account u...@domain in cdrtool the cdrtool try next option(sip domain) if sip domain does not exist it try next option(source ip)and then the default option please let me know if i am wrong This is correct. 2. Determination of the destination id a. CanonicalURI (the destination after all lookups inside the SIP Proxy) b. SipTranslatedRequestURI (the Request URI as presented by the SIP UA) c. CalledStationId (the content of the To header, used as a last resort) if the CanonicalURI does not exist
Re: [OpenSIPS-Users] how to add new domain in CDRTool
Hi Adrian got it Thank you - Original Message From: Adrian Georgescu a...@ag-projects.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thu, November 18, 2010 3:42:35 AM Subject: Re: [OpenSIPS-Users] how to add new domain in CDRTool An example for a valid domain name is 'example.com'. You have tried to add an IP address, which is not the same as a domain name. Adrian On Nov 10, 2010, at 6:46 PM, ha do wrote: hi all how can i add new domain in CDRTool on Rating menu because when i add new domain the CDRTool always said error Error: value '192.168.1.41' for field 'Domain' must be of format 'example.com' Thank you ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need help on connection CDRTool to opensips
Hi Tijimen sorry for late reply 2 opensips2 is a gateway endpoint-opensips1---opensips2internet-VOIP service provider So basically opensips2 is just a gateway to opensips1? [ha] yes, i just try to see what happen in cdrtool, call from subscriber to subscriber these are the logs Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000 callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39 Gateway=192.168.1.36 To=sip:00842...@192.168.1.39 Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited Type=postpaid callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39 log of call-control Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to sip:00842...@192.168.1.39 is postpaid not limited it always return postpaid You need to specify that a sip account is prepaid if you want it to act like a prepaid account. [ha] it is done and work perfectly except 1 thing Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com started for maximum 60 seconds Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com connected for maximum 60 seconds callcontrol.opensips.UNIXSocketProtocol starting on '/var/run/callcontrol/opensips_01.sock' error: cannot write request to /tmp: Connection refused error: failed to end dialog: Cannot send request to OpenSIPS Call id NTJjNzY1NzI1MzgwNmQ2MmU4NDMxMWIzZDgzNWFkOTI. of 1...@cucku.com to sip:00842...@cucku.com disconnected by call control after 60 seconds, call price is 0.0998 prepaid account 100 has only 60s to connect the call, and after 60s callcontrol cannot disconnected call the mi_fifo located in /tmp and opensips.cfg with fifo config # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0600) modparam(mi_fifo, fifo_group, 0) modparam(mi_fifo, fifo_group, root) modparam(mi_fifo, fifo_user, 0) modparam(mi_fifo, fifo_user, root) and the callcontrol config.ini [CallControl] group = root [OpenSIPS] socket_path = /tmp ;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100 terminated i need to increase to 100, right??? how to fix the issue Thank you Ha` - Original Message From: Tijmen de Mes tij...@ag-projects.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Mon, November 15, 2010 4:11:20 PM Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips Hi, The following steps are performed to rate a CDR: 1. Determination of the billing party a. SIP account u...@domain b. SIP domain of the SIP account c. Source IP of the session d. Default (when none of the above matches) if there is no sip account u...@domain in cdrtool the cdrtool try next option(sip domain) if sip domain does not exist it try next option(source ip)and then the default option please let me know if i am wrong This is correct. 2. Determination of the destination id a. CanonicalURI (the destination after all lookups inside the SIP Proxy) b. SipTranslatedRequestURI (the Request URI as presented by the SIP UA) c. CalledStationId (the content of the To header, used as a last resort) if the CanonicalURI does not exist it try to next option(Request URI) ... please let me know if i am wrong This is also correct. 3. Determination of the costs 2 opensips2 is a gateway endpoint-opensips1---opensips2internet-VOIP service provider So basically opensips2 is just a gateway to opensips1? how can i config cdrtool rating all calls from opensips1 and opensips2 could you please guide me some steps to do Did you load the sample data so you can see if a call gets rated? i just try to see what happen in cdrtool, call from subscriber to subscriber these are the logs Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000 callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39 Gateway=192.168.1.36 To=sip:00842...@192.168.1.39 Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited Type=postpaid callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39 log of call-control Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to sip:00842...@192.168.1.39 is postpaid not limited it always return postpaid You need to specify that a sip account is prepaid if you want it to act like a prepaid account. -- Tijmen de Mes AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need help on connection CDRTool to opensips
Hi Tijmen
Maybe you forgot to remove the bind interfaces directive in the mysql config.
that is useful information, i edit the my.cnf and now it works fine now
i read the RATING.txt, PREPAID.txt times to times but i cannot understand it
The following steps are performed to rate a CDR:
1. Determination of the billing party
a. SIP account u...@domain
b. SIP domain of the SIP account
c. Source IP of the session
d. Default (when none of the above matches)
if there is no sip account u...@domain in cdrtool the cdrtool try next
option(sip domain) if sip domain does not exist it try next option(source
ip)and
then the default option
please let me know if i am wrong
2. Determination of the destination id
a. CanonicalURI (the destination after all lookups inside the SIP Proxy)
b. SipTranslatedRequestURI (the Request URI as presented by the SIP UA)
c. CalledStationId (the content of the To header, used as a last resort)
if the CanonicalURI does not exist it try to next option(Request URI) ...
please let me know if i am wrong
3. Determination of the costs
i try to test
1. opensips1 is the registrar server and try to do prepaid for every
subscribers
of opensips
subscriber registers to opensips1, opensips1 domain 192.168.1.39
2 opensips2 is a gateway
endpoint-opensips1---opensips2internet-VOIP service provider
how can i config cdrtool rating all calls from opensips1 and opensips2
could you please guide me some steps to do
i did check the global.inc and cdr_generic.php
-global.inc is a copy of global.inc.simple.sample and i change
# Normalize engine settings
$CDRTool['normalize']['defaultCountryCode'] = 84;
-cdr-generic.php does not change
i config in web
the destination
Ops,Reseller,Trusted
peer,Domain,Subscriber,Destination,Region,Description,Incr,Min Dur,Max Dur,Max
Price
2,084,,vietnam,6,6,0,
rate
Ops,Reseller,Rate,Destination,App,Connect,Duration,Conn In,Duration In
2,0,84,84,audio,0,998,,
profile
Ops,Reseller,Profile,Rate 1,00-H1,Rate 2,H1-H2,Rate 3,H2-H3,Rate 4,H3-24
2,0,84,84,24,0,0,,0,,0
i just try to see what happen in cdrtool, call from subscriber to subscriber
these are the logs
Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000
callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39
Gateway=192.168.1.36 To=sip:00842...@192.168.1.39
Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited Type=postpaid
callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39
log of call-control
Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to
sip:00842...@192.168.1.39 is postpaid not limited
it always return postpaid
Thank you
- Original Message
From: Tijmen de Mes tij...@ag-projects.com
To: OpenSIPS users mailling list users@lists.opensips.org
Sent: Thu, November 11, 2010 4:42:49 PM
Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips
Hi,
You should verify by hand that you can connect to the database by
executing a mysql -h 192.168.1.39 -u opensips -psips opensips.
Maybe you forgot to remove the bind interfaces directive in the mysql config.
Best regards,
Tijmen de Mes
AG Projects
On Nov 9, 2010, at 4:11 AM, ha do wrote:
Hi all
i installed opensips and mediaproxy in server 1, ip address : 192.168.1.39
and cdrtool and freeradius in server 2, ip address : 192.168.1.42
the opensips, mediaproxy, freeradius starting are fine
i use the link below as a guide
http://cdrtool.ag-projects.com/browser/doc/INSTALL.txt
the problem i got only with cdrtool
Nov 9 04:56:22 cdrtool cdrtool[3219]: Read 12291 PSTN destinations from cache
in 0 seconds
Nov 9 04:56:22 cdrtool cdrtool[3219]: Error in connect(): pconnect as
opensips
to 192.168.1.39 failed
Nov 9 04:56:22 cdrtool cdrtool[3219]: Database DB_opensips error: (0)
select
*
from domain
Nov 9 04:56:22 cdrtool cdrtool[3219]: Error in connect(): pconnect as
opensips
to 192.168.1.39 failed
Nov 9 04:56:22 cdrtool cdrtool[3219]: Database DB_opensips error: (0)
select
*
from trusted
Nov 9 04:56:22 cdrtool cdrtool[3219]: PSTN destinations cache size: 0.64 MB
Nov 9 04:56:22 cdrtool cdrtool[3219]: SIP destinations cache size: 0.00 MB
Nov 9 04:56:22 cdrtool cdrtool[3219]: Cached 12291 total, 12291 default, 0
gateway, 0 domain, 0 subscriber destinations
Nov 9 04:56:22 cdrtool cdrtool[3219]: Cached 0 SIP destinations
the cdrtool is fine with almost component except the connection to opensips
the global.inc with define opensips server
class DB_opensips extends DB_Sql {
var $Host = 192.168.1.39;
var $Database = opensips;
var $User = opensips;
var $Password = sips;
var $Halt_On_Error = no;
}
and issue the command grant all on opensips.* to opensips@'192.168.1.%'
identified by 'sips'; in opensips server
please help
Ha`
___
Users mailing list
Users@lists.opensips.org
[OpenSIPS-Users] how to add new domain in CDRTool
hi all how can i add new domain in CDRTool on Rating menu because when i add new domain the CDRTool always said error Error: value '192.168.1.41' for field 'Domain' must be of format 'example.com' Thank you ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] need help on CDRTool
hi
i try to learn CDRTool,
i config the global.inc with
# Normalize engine settings
$CDRTool['normalize']['defaultCountryCode'] = 84;
the cdr_generic.php
class CDRS {
var $CDR_class = 'CDR';
var $intAccessCode = '00';
var $natAccessCode = '0';
var $maxrowsperpage = 15;
var $status = array();
var $normalizedField = 'Normalized';
var $DestinationIdField = 'DestinationId';
var $BillingIdField = 'UserName';
i config in web
the destination
Ops,Reseller,Trusted
peer,Domain,Subscriber,Destination,Region,Description,Incr,Min Dur,Max Dur,Max
Price
2,084,,vietnam,6,6,0,
rate
Ops,Reseller,Rate,Destination,App,Connect,Duration,Conn In,Duration In
2,0,84,84,audio,0,998,,
profile
Ops,Reseller,Profile,Rate 1,00-H1,Rate 2,H1-H2,Rate 3,H2-H3,Rate 4,H3-24
2,0,84,84,24,0,0,,0,,0
when i make call and get the syslog of cdrtool
Nov 11 06:20:57 opensips cdrtool[2312]: MaxSessionTime Duration=36000
callid=1289440810-2424-hiep...@192.168.1.36 From=sip:843...@192.168.1.41
Gateway=192.168.1.36 To=sip:00842...@192.168.1.41
Nov 11 06:20:57 opensips cdrtool[2312]: MaxSessionTime=unlimited Type=postpaid
callid=1289440810-2424-hiep...@192.168.1.36 billingparty=843...@192.168.1.41
get the log from callcontrol
Call id 1289440810-2424-hiep...@192.168.1.36 of 843...@192.168.1.41 to
sip:00842...@192.168.1.41 is postpaid not limited
what do i miss thing to make CDRTool rating a call, please help
Thank you
Ha`
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] need help on connection CDRTool to opensips
Hi all
i installed opensips and mediaproxy in server 1, ip address : 192.168.1.39
and cdrtool and freeradius in server 2, ip address : 192.168.1.42
the opensips, mediaproxy, freeradius starting are fine
i use the link below as a guide
http://cdrtool.ag-projects.com/browser/doc/INSTALL.txt
the problem i got only with cdrtool
Nov 9 04:56:22 cdrtool cdrtool[3219]: Read 12291 PSTN destinations from cache
in 0 seconds
Nov 9 04:56:22 cdrtool cdrtool[3219]: Error in connect(): pconnect as opensips
to 192.168.1.39 failed
Nov 9 04:56:22 cdrtool cdrtool[3219]: Database DB_opensips error: (0) select
*
from domain
Nov 9 04:56:22 cdrtool cdrtool[3219]: Error in connect(): pconnect as opensips
to 192.168.1.39 failed
Nov 9 04:56:22 cdrtool cdrtool[3219]: Database DB_opensips error: (0) select
*
from trusted
Nov 9 04:56:22 cdrtool cdrtool[3219]: PSTN destinations cache size: 0.64 MB
Nov 9 04:56:22 cdrtool cdrtool[3219]: SIP destinations cache size: 0.00 MB
Nov 9 04:56:22 cdrtool cdrtool[3219]: Cached 12291 total, 12291 default, 0
gateway, 0 domain, 0 subscriber destinations
Nov 9 04:56:22 cdrtool cdrtool[3219]: Cached 0 SIP destinations
the cdrtool is fine with almost component except the connection to opensips
the global.inc with define opensips server
class DB_opensips extends DB_Sql {
var $Host = 192.168.1.39;
var $Database = opensips;
var $User = opensips;
var $Password = sips;
var $Halt_On_Error = no;
}
and issue the command grant all on opensips.* to opensips@'192.168.1.%'
identified by 'sips'; in opensips server
please help
Ha`
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] cdrtool install problem
HI Tijmen i insert 2 lines in source.list and it is fine deb http://ftp.us.debian.org/debian/ testing main deb-src http://ftp.us.debian.org/debian/ testing main Thank you Ha - Original Message From: ha do haloha...@yahoo.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Wed, November 3, 2010 8:09:42 AM Subject: Re: [OpenSIPS-Users] cdrtool install problem Hi Tijmen here is the source.list cdrtool:~# uname -a Linux cdrtool 2.6.26-2-686 #1 SMP Thu Sep 16 19:35:51 UTC 2010 i686 GNU/Linux cdrtool:~# cat /etc/apt/sources.list # # deb cdrom:[Debian GNU/Linux 5.0.6 _Lenny_ - Official i386 CD Binary-1 20100904-18:58]/ lenny main deb cdrom:[Debian GNU/Linux 5.0.6 _Lenny_ - Official i386 CD Binary-1 20100904-18:58]/ lenny main deb http://ftp.us.debian.org/debian/ lenny main deb-src http://ftp.us.debian.org/debian/ lenny main deb http://ag-projects.com/debian unstable main deb-src http://ag-projects.com/debian unstable main deb http://security.debian.org/ lenny/updates main deb-src http://security.debian.org/ lenny/updates main deb http://volatile.debian.org/debian-volatile lenny/volatile main deb-src http://volatile.debian.org/debian-volatile lenny/volatile main let me try on Debian unstable Thank you Ha - Original Message From: Tijmen de Mes tij...@ag-projects.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Wed, November 3, 2010 2:38:10 AM Subject: Re: [OpenSIPS-Users] cdrtool install problem Hi, I am unable to reproduce this,. I can install php5-geoip, geoip- database and cdrtool fine on Debian unstable. Which version of Debian are you using? Best regards, Tijmen de Mes AG Projects On Nov 2, 2010, at 4:03 PM, ha do wrote: Hi i try to install cdrtool in debian OS but i got stuck when install cdrtool cdrtool:/mnt# apt-get install php5-geoip Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: libgeoip1 Suggested packages: geoip-bin The following packages will be REMOVED: geoip-database The following NEW packages will be installed: libgeoip1 php5-geoip 0 upgraded, 2 newly installed, 1 to remove and 0 not upgraded. Need to get 0B/660kB of archives. After this operation, 4403kB disk space will be freed. Do you want to continue [Y/n]? cdrtool requires php5-geoip + geoip-database but the php5-geoip dependent on libgeoip1. when i install libgeoip1 it remove geoip-database Please help Thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] cdrtool install problem
Hi i try to install cdrtool in debian OS but i got stuck when install cdrtool cdrtool:/mnt# apt-get install php5-geoip Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: libgeoip1 Suggested packages: geoip-bin The following packages will be REMOVED: geoip-database The following NEW packages will be installed: libgeoip1 php5-geoip 0 upgraded, 2 newly installed, 1 to remove and 0 not upgraded. Need to get 0B/660kB of archives. After this operation, 4403kB disk space will be freed. Do you want to continue [Y/n]? cdrtool requires php5-geoip + geoip-database but the php5-geoip dependent on libgeoip1. when i install libgeoip1 it remove geoip-database Please help Thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] cdrtool install problem
Hi Tijmen here is the source.list cdrtool:~# uname -a Linux cdrtool 2.6.26-2-686 #1 SMP Thu Sep 16 19:35:51 UTC 2010 i686 GNU/Linux cdrtool:~# cat /etc/apt/sources.list # # deb cdrom:[Debian GNU/Linux 5.0.6 _Lenny_ - Official i386 CD Binary-1 20100904-18:58]/ lenny main deb cdrom:[Debian GNU/Linux 5.0.6 _Lenny_ - Official i386 CD Binary-1 20100904-18:58]/ lenny main deb http://ftp.us.debian.org/debian/ lenny main deb-src http://ftp.us.debian.org/debian/ lenny main deb http://ag-projects.com/debian unstable main deb-src http://ag-projects.com/debian unstable main deb http://security.debian.org/ lenny/updates main deb-src http://security.debian.org/ lenny/updates main deb http://volatile.debian.org/debian-volatile lenny/volatile main deb-src http://volatile.debian.org/debian-volatile lenny/volatile main let me try on Debian unstable Thank you Ha - Original Message From: Tijmen de Mes tij...@ag-projects.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Wed, November 3, 2010 2:38:10 AM Subject: Re: [OpenSIPS-Users] cdrtool install problem Hi, I am unable to reproduce this,. I can install php5-geoip, geoip- database and cdrtool fine on Debian unstable. Which version of Debian are you using? Best regards, Tijmen de Mes AG Projects On Nov 2, 2010, at 4:03 PM, ha do wrote: Hi i try to install cdrtool in debian OS but i got stuck when install cdrtool cdrtool:/mnt# apt-get install php5-geoip Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: libgeoip1 Suggested packages: geoip-bin The following packages will be REMOVED: geoip-database The following NEW packages will be installed: libgeoip1 php5-geoip 0 upgraded, 2 newly installed, 1 to remove and 0 not upgraded. Need to get 0B/660kB of archives. After this operation, 4403kB disk space will be freed. Do you want to continue [Y/n]? cdrtool requires php5-geoip + geoip-database but the php5-geoip dependent on libgeoip1. when i install libgeoip1 it remove geoip-database Please help Thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] problem with opensips startup on Centos 5
Hi all i think the script file in packaging/fedora/opensips.init is not create the /var/run/opensips.pid it is the same problem with Chrispy? — 01 September 2010, 01:46on webpage http://www.opensips.org/Resources/DocsTsStart for workaround : service opensips stop then issue : opensipsctl start - /var/run/opensips.pid should create successfull but when server restart i cannot use the munin service could someone help me Thank you Ha` - Original Message From: ha do haloha...@yahoo.com To: OpenSIPS users@lists.opensips.org Sent: Tue, September 14, 2010 10:05:20 AM Subject: [OpenSIPS-Users] problem with opensips startup on Centos 5 Hello all i use Centos 5: [r...@localhost ~]# uname -a Linux localhost.localdomain 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:35 EDT 2010 i686 i686 i386 GNU/Linux i am testing the opensips 1.6.2 and i installed successfully opensips the step to get problem: 1. Install opensips successfull 2. copy file packaging/fedora/opensips.init to /etc/init.d/opensips 3. chmod 755 /etc/init.d/opensips 4. chkconfig --add /etc/init.d/opensips 5. restart Server 6. check the opensips status: [r...@localhost ~]# ps -ef | grep opensips root 1539 1 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1543 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1544 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1545 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1546 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1547 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1548 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1574 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1575 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1576 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1577 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1578 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1579 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1599 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1600 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1601 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1602 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1650 1627 0 22:57 pts/000:00:00 grep opensips 7. i use the opensipsctl restart and got the problem : [r...@localhost ~]# opensipsctl restart INFO: Restarting OpenSIPS : ERROR: No PID file found (/var/run/opensips.pid)! OpenSIPS probably not running INFO: Starting OpenSIPS : ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start failed could someone help me, i want opensips startup along with Server and the opensipsctl command should work properly Thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] get error with opensip-cp on centos 5
if someone else got the same problem with me, to solve the issue please follow
the link
http://lists.opensips.org/pipermail/users/2010-June/013139.html
Thank you
Ha`
- Original Message
From: ha do haloha...@yahoo.com
To: OpenSIPS users@lists.opensips.org
Sent: Tue, September 14, 2010 11:12:47 AM
Subject: [OpenSIPS-Users] get error with opensip-cp on centos 5
Hi all
i try to use the opensips-cp 4:
i login the the web page http://192.168.1.100/cp successfull
the menu Admin + User are working properly
but the system menu does not work
i get the error on the webpage:
sorry -- cannot open write fifo
and this is the error in debug message :
Sep 14 00:10:42 localhost kernel: type=1400 audit(1284437442.630:49): avc:
denied { getattr } for pid=2093 comm=httpd path=/tmp/opensips_fifo
dev=dm-0 ino=884744 scontext=root:system_r:httpd_t:s0
tcontext=root:object_r:tmp_t:s0 tclass=fifo_file
i did set fifo_mode, 0666 in opensips.cfg
please help,what should i do to resolve problem
Thank you
Ha`
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] problem with opensips startup on Centos 5
Hello all i use Centos 5: [r...@localhost ~]# uname -a Linux localhost.localdomain 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:35 EDT 2010 i686 i686 i386 GNU/Linux i am testing the opensips 1.6.2 and i installed successfully opensips the step to get problem: 1. Install opensips successfull 2. copy file packaging/fedora/opensips.init to /etc/init.d/opensips 3. chmod 755 /etc/init.d/opensips 4. chkconfig --add /etc/init.d/opensips 5. restart Server 6. check the opensips status: [r...@localhost ~]# ps -ef | grep opensips root 1539 1 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1543 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1544 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1545 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1546 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1547 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1548 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1574 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1575 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1576 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1577 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1578 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1579 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1599 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1600 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1601 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1602 1539 0 22:56 ?00:00:00 /usr/local/sbin/opensips root 1650 1627 0 22:57 pts/000:00:00 grep opensips 7. i use the opensipsctl restart and got the problem : [r...@localhost ~]# opensipsctl restart INFO: Restarting OpenSIPS : ERROR: No PID file found (/var/run/opensips.pid)! OpenSIPS probably not running INFO: Starting OpenSIPS : ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start failed could someone help me, i want opensips startup along with Server and the opensipsctl command should work properly Thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] get error with opensip-cp on centos 5
Hi all
i try to use the opensips-cp 4:
i login the the web page http://192.168.1.100/cp successfull
the menu Admin + User are working properly
but the system menu does not work
i get the error on the webpage:
sorry -- cannot open write fifo
and this is the error in debug message :
Sep 14 00:10:42 localhost kernel: type=1400 audit(1284437442.630:49): avc:
denied { getattr } for pid=2093 comm=httpd path=/tmp/opensips_fifo
dev=dm-0 ino=884744 scontext=root:system_r:httpd_t:s0
tcontext=root:object_r:tmp_t:s0 tclass=fifo_file
i did set fifo_mode, 0666 in opensips.cfg
please help,what should i do to resolve problem
Thank you
Ha`
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0
Hi Alex i try new opensips-cp and it work fine now but i have problem with add new admin in opensips-cp webpage step1: add new admin with name : cucku step2: add successfull step3: login with username cucku -- the issue is here, there is no available tools for new admin which is added via webpage the available tool for new admin is absent :(,where do i need to add available tool for admin with name : cucku the query from mysql : mysql select * from ocp_admin_privileges\G; *** 1. row *** id: 55 first_name: last_name: username: admin password: admin ha1: d2abaa37a7c3db1137d385e1d8c15fd2 available_tools: all permissions: all *** 2. row *** id: 56 first_name: cucku last_name: cucku username: cucku password: ha1: 91730b420f600d8da7817d96c20a59b9 available_tools: permissions: NULL 2 rows in set (0.00 sec) ERROR: No query specified Thank you Ha` --- On Thu, 3/18/10, Alex Ionescu a...@opensips.org wrote: From: Alex Ionescu a...@opensips.org Subject: Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0 To: OpenSIPS users mailling list users@lists.opensips.org Date: Thursday, March 18, 2010, 5:46 AM Hi, Actually there were a few issues that were found and solved the last few days. I hope you won't have anymore problems if you download the new 4.0 tarball. So, give it a try and let me know. Regards, Alex ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0
HI Alex
did you find new issue on opensips cp
Thank youHa`
--- On Sat, 3/13/10, Alex Ionescu a...@opensips.org wrote:
From: Alex Ionescu a...@opensips.org
Subject: Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Saturday, March 13, 2010, 9:21 AM
Hi,
I will investigate the problem and will let you know if I find any
issue.
Regards,
Alex
On 3/13/2010 06:39, Do Nguyen Ha wrote:
Hi Alex
here is mysql query:
mysql select * from ocp_admin_privileges\G;
*** 1. row ***
id: 58
first_name:
last_name:
username: admin
password: admin
ha1: d2abaa37a7c3db1137d385e1d8c15fd2
available_tools: all
permissions: all
so i am still getting the invalid username on opensips-cp version 4
the i remove the folder opensips-cp version 4 and replace with
opensips-cp version 3 and now it works without problem with opensips-cp
version 3
i dont change anything and step i do :
step 1. cd /var/www
step 2. delete folder opensips-cp
step 3. download the opensips-cp_3.0.tar.gz
step 4. untar the file opensips-cp_3.0
step 5. chown www-data:www-data opensips-cp -R
Thank you for your help
Ha
--- On Fri, 3/12/10, Alex Ionescu a...@opensips.org
wrote:
From: Alex Ionescu a...@opensips.org
Subject: Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Friday, March 12, 2010, 9:02 AM
Sorry,
My mistake ... the MD5 should be applied on the value not on the field
name ... So, it is :
INSERT INTO ocp_admin_privileges
(username,password,ha1,available_tools,permissions) values
('admin','admin',md5('admin:admin'),'all','all');
On 3/12/2010 13:06, Alex Ionescu wrote:
Hi,
The problem seems to be your step 11 :
step 11. INSERT INTO ocp_admin_privileges
(username,password,ha1,available_tools,permissions) values
('admin','admin','admin:admin','all','all');
You must encode the ha1 field. So, I believe that the correct
query would be :
INSERT INTO ocp_admin_privileges
(username,password,md5(ha1),available_tools,permissions) values
('admin','admin','admin:admin','all','all');
Anyway, you can always check the INSTALL file (I think the query is
located on line 102 or 103).
Regards,
Alex
On 3/12/2010 12:41, Do Nguyen Ha wrote:
step 11. INSERT INTO
ocp_admin_privileges
(username,password,ha1,available_tools,permissions) values
('admin','admin','admin:admin','all','all');
--
Alex Ionescu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Ionescu
www.voice-system.ro
-Inline Attachment Follows-
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Ionescu
www.voice-system.ro
-Inline Attachment Follows-
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need help on dialplan module
Hi Bogdan i refer : String translation (regexp detection, subst translation) function the repl_exp = a_value\1 the dialplan will use the a_value + subst_exp as the output if the match_exp=true the repl_exp = a_value\2 the dialplan will use the columm a_value + (input string - subst_exp) as the output if the match_exp=true it is right? Thank you Ha` --- On Fri, 1/22/10, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: From: Bogdan-Andrei Iancu bog...@voice-system.ro Subject: Re: [OpenSIPS-Users] need help on dialplan module To: OpenSIPS users mailling list users@lists.opensips.org Date: Friday, January 22, 2010, 9:55 AM Hi Ha, The modules user PERL like substitution. A fast google gives some docs on this: http://www.anaesthetist.com/mnm/perl/Findex.htm#regex.htm Regards, Bogdan ha do wrote: Hi all could you please need me to understand the translation on dialplan module; mysql select * from dialplan; ++--++--+---+---++--+---+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | ++--++--+---+---++--+---+ | 73 | 15 | 0 | 1 | ^000 | 0 | ^(0)(.+) | \2 | | | 78 | 16 | 0 | 1 | 000 | 0 | (000)(.+) | 8\2 | | | 76 | 14 | 0 | 1 | ^000 | 0 | ^(000)(.+) | 8\2 | | | 75 | 15 | 0 | 1 | ^55 | 0 | ^(55)(.+) | \2 | | ++--++--+---+---++--+---+ [r...@localhost ~]# opensipsctl fifo dp_translate 14 00055980007 Output:: 855980007 [r...@localhost ~]# opensipsctl fifo dp_translate 15 0007 Output:: 007 [r...@localhost ~]# opensipsctl fifo dp_translate 15 55980007 Output:: 980007 [r...@localhost ~]# opensipsctl fifo dp_translate 16 55980007 Output:: 87 repl_exp : sometimes has value \2 or \1 - what does it mean?? does it have other value? what does the ^ mean?? is there more special character?? where do i find more docs for translation rule Thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] need help on dialplan module
Hi all could you please need me to understand the translation on dialplan module; mysql select * from dialplan; ++--++--+---+---++--+---+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | ++--++--+---+---++--+---+ | 73 | 15 | 0 | 1 | ^000 | 0 | ^(0)(.+) | \2 | | | 78 | 16 | 0 | 1 | 000 | 0 | (000)(.+) | 8\2 | | | 76 | 14 | 0 | 1 | ^000 | 0 | ^(000)(.+) | 8\2 | | | 75 | 15 | 0 | 1 | ^55 | 0 | ^(55)(.+) | \2 | | ++--++--+---+---++--+---+ [r...@localhost ~]# opensipsctl fifo dp_translate 14 00055980007 Output:: 855980007 [r...@localhost ~]# opensipsctl fifo dp_translate 15 0007 Output:: 007 [r...@localhost ~]# opensipsctl fifo dp_translate 15 55980007 Output:: 980007 [r...@localhost ~]# opensipsctl fifo dp_translate 16 55980007 Output:: 87 repl_exp : sometimes has value \2 or \1 - what does it mean?? does it have other value? what does the ^ mean?? is there more special character?? where do i find more docs for translation rule Thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need help on mediaproxy ports
Thank you for the link Ha` --- On Tue, 1/19/10, Duane Larson duane.lar...@gmail.com wrote: From: Duane Larson duane.lar...@gmail.com Subject: Re: [OpenSIPS-Users] need help on mediaproxy ports To: OpenSIPS users mailling list users@lists.opensips.org Date: Tuesday, January 19, 2010, 8:11 AM I believe you will always have (x ports * 2) since RTP is always accompanied by an odd numbered RTCP port. So you do have only 2 RTP ports listed above, but those RTP ports also require a RTCP port. Look at this http://en.wikipedia.org/wiki/RTP_Control_Protocol On Tue, Jan 19, 2010 at 1:56 AM, ha do haloha...@yahoo.com wrote: Hi all i debug on mediaproxy and see, mediaproxy create 4 ports but the real ports to relay media are only 2 ports: 118.69.239.140:50012 - 118.69.239.140:50014 is it normal, can i config mediaproxy create only 2 ports Thank you Ha` mediaproxy.mediacontrol.StreamListenerProtocol starting on 50012 mediaproxy.mediacontrol.StreamListenerProtocol starting on 50013 mediaproxy.mediacontrol.StreamListenerProtocol starting on 50014 mediaproxy.mediacontrol.StreamListenerProtocol starting on 50015 debug: Added new stream: (audio) 192.168.1.4:45746 (RTP: Unknown, RTCP: Unknown) - 118.69.239.140:50012 - 118.69.239.140:50014 - Unknown (RTP: Unknown, RTCP: Unknown) debug: created new session 7810dcdd0ceb7...@192.168.1.4: 8...@118.69.239.140 (93846fc44ae3fe24) -- 9...@118.69.239.140 debug: updating existing session 7810dcdd0ceb7...@192.168.1.4: 8...@118.69.239.140 (93846fc44ae3fe24) -- 9...@118.69.239.140 debug: Received updated SDP answer debug: Got initial answer from callee for stream: (audio) 192.168.1.4:45746 (RTP: Unknown, RTCP: Unknown) - 118.69.239.140:50012 - 118.69.239.140:50014 - 192.168.1.6:48000 (RTP: Unknown, RTCP: Unknown) debug: Got traffic information for stream: (audio) 192.168.1.4:45746 (RTP: 210.245.35.150:45746, RTCP: Unknown) - 118.69.239.140:50012 - 118.69.239.140:50014 - 192.168.1.6:48000 (RTP: Unknown, RTCP: Unknown) debug: Got traffic information for stream: (audio) 192.168.1.4:45746 (RTP: 210.245.35.150:45746, RTCP: Unknown) - 118.69.239.140:50012 - 118.69.239.140:50014 - 192.168.1.6:48000 (RTP: 210.245.35.150:48000, RTCP: Unknown) debug: removing session 7810dcdd0ceb7...@192.168.1.4: 8...@118.69.239.140 (93846fc44ae3fe24) -- 9...@118.69.239.140 (Port 50012 Closed) (Port 50013 Closed) (Port 50014 Closed) (Port 50015 Closed) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -Inline Attachment Follows- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] need help on mediaproxy ports
Hi all i debug on mediaproxy and see, mediaproxy create 4 ports but the real ports to relay media are only 2 ports: 118.69.239.140:50012 - 118.69.239.140:50014 is it normal, can i config mediaproxy create only 2 ports Thank you Ha` mediaproxy.mediacontrol.StreamListenerProtocol starting on 50012 mediaproxy.mediacontrol.StreamListenerProtocol starting on 50013 mediaproxy.mediacontrol.StreamListenerProtocol starting on 50014 mediaproxy.mediacontrol.StreamListenerProtocol starting on 50015 debug: Added new stream: (audio) 192.168.1.4:45746 (RTP: Unknown, RTCP: Unknown) - 118.69.239.140:50012 - 118.69.239.140:50014 - Unknown (RTP: Unknown, RTCP: Unknown) debug: created new session 7810dcdd0ceb7...@192.168.1.4: 8...@118.69.239.140 (93846fc44ae3fe24) -- 9...@118.69.239.140 debug: updating existing session 7810dcdd0ceb7...@192.168.1.4: 8...@118.69.239.140 (93846fc44ae3fe24) -- 9...@118.69.239.140 debug: Received updated SDP answer debug: Got initial answer from callee for stream: (audio) 192.168.1.4:45746 (RTP: Unknown, RTCP: Unknown) - 118.69.239.140:50012 - 118.69.239.140:50014 - 192.168.1.6:48000 (RTP: Unknown, RTCP: Unknown) debug: Got traffic information for stream: (audio) 192.168.1.4:45746 (RTP: 210.245.35.150:45746, RTCP: Unknown) - 118.69.239.140:50012 - 118.69.239.140:50014 - 192.168.1.6:48000 (RTP: Unknown, RTCP: Unknown) debug: Got traffic information for stream: (audio) 192.168.1.4:45746 (RTP: 210.245.35.150:45746, RTCP: Unknown) - 118.69.239.140:50012 - 118.69.239.140:50014 - 192.168.1.6:48000 (RTP: 210.245.35.150:48000, RTCP: Unknown) debug: removing session 7810dcdd0ceb7...@192.168.1.4: 8...@118.69.239.140 (93846fc44ae3fe24) -- 9...@118.69.239.140 (Port 50012 Closed) (Port 50013 Closed) (Port 50014 Closed) (Port 50015 Closed) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need advice on B2b
HI Anca
it is clear now :)
i am trying to learn the opensips,
Thank you
Ha`
--- On Wed, 1/6/10, Anca Vamanu a...@opensips.org wrote:
From: Anca Vamanu a...@opensips.org
Subject: Re: [OpenSIPS-Users] need advice on B2b
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Wednesday, January 6, 2010, 3:14 AM
Hi Ha`,
The excerpt from your script shows that you don't have a good
understanding of the opensips scripting logic. First, the route
block will only be called for SIP Requests.
Calling this is not right:
if(status==200)
route(b2b_reply);
The replies will go into reply route blocks. You can have a default
reply route ( one without an index or with index 0), or you can specify
a certain reply route for a request by calling t_on_reply.
Second, you don't understand what happens with b2b request and replies.
It is explained in the documentation:
The requests and replies that are received by the B2BUA server,
belonging to the dialogs it is handling will not go into the script as
normal request do. The reason for this is that this are not normal
requests where the server is a proxy, but the server is an endpoint in
the dialog and therefore they should not go through the same routes.
However, it is normal for this request to be seen from the script and
allow the script writer to do the processing it desires based on them.
For this, it is possible to define two special B2B routes - one for
requests and one for replies. The routes are of type route
and have their name defined in the modules parameters script_req_route
and script_reply_route.
In other words, there are two important things:
1. the B2B requests/replies will not go into the default request/reply
route block.
2. the b2b_request/b2_reply route will be called automatically for
every request/reply targeted to the b2b agent
So, for your script, you don't need this lines:
if(is_method(INVITE)) {
route(b2b_request);
t_on_reply(2);
}
else
if(status==200)
route(b2b_reply);
Hope this made things a bit clearer.
Regards,
--
Anca Vamanu
www.voice-system.ro
ha do wrote:
HI Anca
i am trying to use the b2b_request + b2b_reply
route{
...
if(is_method(INVITE) !(src_ip == 192.168.1.249
src_port ==5060))
{
if (! t_newtran()){
sl_reply_error();
exit;
};
b2b_init_request(top hiding);
exit;
};
route(1);
}
route[1] {
if(is_method(INVITE)) {
route(b2b_request);
t_on_reply(2);
}
else
if(status==200)
route(b2b_reply);
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[b2b_request] {
xlog(b2b_request cucku ($ci)\n);
force_rtp_proxy();
}
route[b2b_reply] {
xlog(b2b_reply cucku ($ci)\n);
force_rtp_proxy();
}
i get the errors :
ERROR:nathelper:force_rtp_proxy: Unable to parse body
and
DBG:tm:utimer_routine: timer routine:4,tl=0xb615e9a8 next=(nil),
timeout=2900
DBG:tm:timer_routine: timer routine:3,tl=0xb615c6f4 next=(nil),
timeout=29
DBG:tm:delete_handler: removing 0xb615c690
DBG:tm:delete_cell: delete_cell 0xb615c690: can't delete -- still
reffed (1)
===
DBG:core:get_hdr_field: cseq CSeq: 2 INVITE
DBG:core:parse_headers: flags=8
DBG:tm:t_reply_matching: hash 21530 label 76806763 branch 0
DBG:tm:t_reply_matching: REF_UNSAFE: after is 2
DBG:tm:t_reply_matching: reply matched (T=0xb615e85c)!
DBG:tm:t_check: end=0xb615e85c
DBG:tm:reply_received: org. status uas=0, uac[0]=100 local=2
is_invite=1)
DBG:tm:t_should_relay_response: T_code=0, new_code=180
DBG:tm:local_reply: branch=0, save=0, winner=0
DBG:tm:local_reply: Passing provisional reply 180 to FIFO application
DBG:tm:run_trans_callbacks: trans=0xb615e85c, callback type 1024, id 0
entered
DBG:b2b_entities:b2b_parse_key: hash_index = [111] - local_index= [0]
DBG:core:parse_headers: flags=
DBG:core:get_hdr_field: content_length=0
DBG:core:get_hdr_field: found end of header
DBG:b2b_entities:b2b_tm_cback: Received a reply with statuscode = 180
DBG:core:parse_headers: flags=
DBG:b2b_entities:b2b_new_dlg: 'To' header ALREADY PARSED:
sip:1...@192.168.1.249
DBG:b2b_entities:b2b_new_dlg: Not an initial request
DBG:core:parse_to_param: tag=bfad35cdb22f09f741816636d344f54b-19f0
DBG:core:parse_to: end of header reached, state=29
DBG:core:parse_to: display={},
ruri={sip:0873000...@192.168.1.249;user=phone}
DBG:core:print_rr_body: current rr is sip:192.168.1.249;lr=on
DBG:core:print_rr_body: out rr [sip:192.168.1.249;lr
Re: [OpenSIPS-Users] Need help on flag in usrloc
Hi Bogdan
it is clear now :)
Thank you
Ha`
--- On Wed, 1/6/10, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need help on flag in usrloc
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Wednesday, January 6, 2010, 3:51 AM
Hi Ha,
the two flags are different and may have different values - one is used
as NAT marker, the other one is used as SIP-based pinging marker.
so, you can use different flags and both of them will be saved in cflag
mask.
Regards,
Bogdan
ha do wrote:
Hi Bogdan
got it :)
1 more question about the flag
modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 7)
the modem ADSL will close the port after 3 mins(some minutes), so
Opensips should send OPTION message(sip ping) to modem to keep port
that should open for UA
the sipping_blag of nathelper module should be the same value as
nat_bflag of usrloc ? or the cflag of usrloc just has a value??
Thank you
Ha`
--- On *Tue, 1/5/10, Bogdan-Andrei Iancu /bog...@voice-system.ro/*
wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need help on flag in usrloc
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, January 5, 2010, 7:20 AM
Hi Ha,
the NAT branch flag you use is 7 (nat_bflag) and in usrloc you
find the
branch flags in the cflags (contact flags) field. The cflags is a
mask
with all the branch flags: 192 = 128 (2^7) + 64 (2^6)
Regards,
Bogdan
ha do wrote:
Hi all
i am successfull to check the UA behind NAT but i dont know what
value
of the flag will be stored in the usrloc
Could someone please let me know the value of Nated UA flag,
that is
stored in usrloc
my config :
modparam(nathelper, natping_interval,180)
modparam(nathelper, ping_nated_only, 1)
modparam(nathelper, sipping_bflag, 7)
modparam(nathelper, sipping_from, sip:cu...@kamailio.org
/mc/compose?to=cu...@kamailio.org)
modparam(registrar|nathelper, received_avp, $avp(i:80))
modparam(usrloc, nat_bflag, 7)
route{
route(4);
if (method==REGISTER)
{
if (isflagset(5)) {
setbflag(6);
setbflag(7);
}
if (!save(location))
sl_reply_error();
exit;
}
}
route[4]{
force_rport();
if (nat_uac_test(19)) {
if (method==REGISTER) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(5);
}
return;
}
mysql select * from location\G
*** 1. row ***
id: 12
username: 1000
domain: NULL
contact: sip:1...@192.168.1.2:12280;rinstance=a4752f45bdc3ddd3
received: sip:210.245.35.150:12280
path: NULL
expires: 2010-01-05 17:48:56
q: -1.00
callid: YjE1NWJmYWYyYWMzZDg2ZDc5MjY0NDQyMGE5NDEwM2E.
cseq: 2
last_modified: 2010-01-05 16:48:56
flags: 0
cflags: 192
user_agent: eyeBeam release 1004p stamp 31962
socket: udp:118.69.193.198:5060
methods: 5951
Thank you
Ha`
___
Users mailing list
Users@lists.opensips.org /mc/compose?to=us...@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users mailing list
us...@lists.opensips.org /mc/compose?to=us...@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] need help on mediaproxy that cannot start
Hi all i follow the instruction : http://www.smartvox.co.uk/serfaq_install_mediaproxy2.htm my centos version: [r...@centos-cucku application]# uname -a Linux CentOS-Cucku 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux i get the error when start the mediaproxy on Centos : [r...@centos-cucku application]# /usr/bin/media-dispatcher Traceback (most recent call last): File /usr/bin/media-dispatcher, line 12, in ? from application.process import process, ProcessError File /usr/lib/python2.4/site-packages/application/process.py, line 12, in ? from application import log File /usr/lib/python2.4/site-packages/application/log/__init__.py, line 12, in ? from application.log.extensions import twisted File /usr/lib/python2.4/site-packages/application/log/extensions/twisted/__init__.py, line 4 from __future__ import absolute_import SyntaxError: future feature absolute_import is not defined please help thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need advice on B2b
HI Anca
i am trying to use the b2b_request + b2b_reply
route{
...
if(is_method(INVITE) !(src_ip == 192.168.1.249 src_port ==5060))
{
if (! t_newtran()){
sl_reply_error();
exit;
};
b2b_init_request(top hiding);
exit;
};
route(1);
}
route[1] {
if(is_method(INVITE)) {
route(b2b_request);
t_on_reply(2);
}
else
if(status==200)
route(b2b_reply);
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[b2b_request] {
xlog(b2b_request cucku ($ci)\n);
force_rtp_proxy();
}
route[b2b_reply] {
xlog(b2b_reply cucku ($ci)\n);
force_rtp_proxy();
}
i get the errors :
ERROR:nathelper:force_rtp_proxy: Unable to parse body
and
DBG:tm:utimer_routine: timer routine:4,tl=0xb615e9a8 next=(nil),
timeout=2900
DBG:tm:timer_routine: timer routine:3,tl=0xb615c6f4 next=(nil), timeout=29
DBG:tm:delete_handler: removing 0xb615c690
DBG:tm:delete_cell: delete_cell 0xb615c690: can't delete -- still reffed (1)
===
DBG:core:get_hdr_field: cseq CSeq: 2 INVITE
DBG:core:parse_headers: flags=8
DBG:tm:t_reply_matching: hash 21530 label 76806763 branch 0
DBG:tm:t_reply_matching: REF_UNSAFE: after is 2
DBG:tm:t_reply_matching: reply matched (T=0xb615e85c)!
DBG:tm:t_check: end=0xb615e85c
DBG:tm:reply_received: org. status uas=0, uac[0]=100 local=2 is_invite=1)
DBG:tm:t_should_relay_response: T_code=0, new_code=180
DBG:tm:local_reply: branch=0, save=0, winner=0
DBG:tm:local_reply: Passing provisional reply 180 to FIFO application
DBG:tm:run_trans_callbacks: trans=0xb615e85c, callback type 1024, id 0 entered
DBG:b2b_entities:b2b_parse_key: hash_index = [111] - local_index= [0]
DBG:core:parse_headers: flags=
DBG:core:get_hdr_field: content_length=0
DBG:core:get_hdr_field: found end of header
DBG:b2b_entities:b2b_tm_cback: Received a reply with statuscode = 180
DBG:core:parse_headers: flags=
DBG:b2b_entities:b2b_new_dlg: 'To' header ALREADY PARSED:
sip:1...@192.168.1.249
DBG:b2b_entities:b2b_new_dlg: Not an initial request
DBG:core:parse_to_param: tag=bfad35cdb22f09f741816636d344f54b-19f0
DBG:core:parse_to: end of header reached, state=29
DBG:core:parse_to: display={}, ruri={sip:0873000...@192.168.1.249;user=phone}
DBG:core:print_rr_body: current rr is sip:192.168.1.249;lr=on
DBG:core:print_rr_body: out rr [sip:192.168.1.249;lr=on]
DBG:core:print_rr_body: we have 1 records
DBG:b2b_entities:b2b_tm_cback: Created new dialog structure 0xb61618c0
DBG:core:print_rr_body: current rr is sip:192.168.1.249;lr=on
DBG:core:print_rr_body: out rr [sip:192.168.1.249;lr=on]
DBG:core:print_rr_body: we have 1 records
DBG:b2b_logic:b2bl_parse_key: hash_index = [623] - local_index= [0]
DBG:core:parse_headers: flags=
DBG:b2b_entities:b2b_parse_key: hash_index = [346] - local_index= [0]
DBG:core:parse_headers: flags=
DBG:core:check_ip_address: params 192.168.1.4, 192.168.1.4, 0
DBG:tm:t_reply_with_body: buffer computed
DBG:tm:_reply_light: reply sent out. buf=0x81c70b8: SIP/2.0 1...,
shmem=0xb615e534: SIP/2.0 1
DBG:tm:_reply_light: finished
b2b_reply cucku (B2B.111.0.1262765386)
DBG:core:parse_headers: flags=
DBG:core:parse_headers: flags=1000
DBG:core:parse_content_type_hdr: missing Content-Type header
ERROR:nathelper:force_rtp_proxy: Unable to parse body
Thank you
Ha`
--- On Mon, 1/4/10, Anca Vamanu a...@opensips.org wrote:
From: Anca Vamanu a...@opensips.org
Subject: Re: [OpenSIPS-Users] need advice on B2b
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Monday, January 4, 2010, 3:04 AM
Hi Ha`,
There is a very simple example in the documentation:
route[b2b_request] {
xlog(b2b_request ($ci)\n);
}
route[b2b_reply] {
xlog(b2b_reply ($ci)\n);
}
You can call in these routes any function that you call in a request route.
Regards,
--
Anca Vamanu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Need help on flag in usrloc
Hi Bogdan
got it :)
1 more question about the flag
modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 7)
the modem ADSL will close the port after 3 mins(some minutes), so Opensips
should send OPTION message(sip ping) to modem to keep port that should open for
UA
the sipping_blag of nathelper module should be the same value as nat_bflag of
usrloc ? or the cflag of usrloc just has a value??
Thank you
Ha`
--- On Tue, 1/5/10, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need help on flag in usrloc
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, January 5, 2010, 7:20 AM
Hi Ha,
the NAT branch flag you use is 7 (nat_bflag) and in usrloc you find the
branch flags in the cflags (contact flags) field. The cflags is a mask
with all the branch flags: 192 = 128 (2^7) + 64 (2^6)
Regards,
Bogdan
ha do wrote:
Hi all
i am successfull to check the UA behind NAT but i dont know what value
of the flag will be stored in the usrloc
Could someone please let me know the value of Nated UA flag, that is
stored in usrloc
my config :
modparam(nathelper, natping_interval,180)
modparam(nathelper, ping_nated_only, 1)
modparam(nathelper, sipping_bflag, 7)
modparam(nathelper, sipping_from, sip:cu...@kamailio.org)
modparam(registrar|nathelper, received_avp, $avp(i:80))
modparam(usrloc, nat_bflag, 7)
route{
route(4);
if (method==REGISTER)
{
if (isflagset(5)) {
setbflag(6);
setbflag(7);
}
if (!save(location))
sl_reply_error();
exit;
}
}
route[4]{
force_rport();
if (nat_uac_test(19)) {
if (method==REGISTER) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(5);
}
return;
}
mysql select * from location\G
*** 1. row ***
id: 12
username: 1000
domain: NULL
contact: sip:1...@192.168.1.2:12280;rinstance=a4752f45bdc3ddd3
received: sip:210.245.35.150:12280
path: NULL
expires: 2010-01-05 17:48:56
q: -1.00
callid: YjE1NWJmYWYyYWMzZDg2ZDc5MjY0NDQyMGE5NDEwM2E.
cseq: 2
last_modified: 2010-01-05 16:48:56
flags: 0
cflags: 192
user_agent: eyeBeam release 1004p stamp 31962
socket: udp:118.69.193.198:5060
methods: 5951
Thank you
Ha`
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] something on stun documment
Hi admin http://www.opensips.org/html/docs/modules/devel/stun.html#id227269 1.3.3. alternate_ip (str) Another ip from another interface. Example 1.3. Set alternate_ip parameter ... modparam(stun,alternate_port,3479) -- this is right? Thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Need help Nathelper + rtpproxy
Hi Bogdan
Thank you for the tip :)
--- On Tue, 12/29/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need help Nathelper + rtpproxy
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, December 29, 2009, 7:42 AM
Hi Ha,
You need to call unforce_rtp_proxy() when BYE is received.
Regards,
Bogdan
# - request routing logic ---
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
};
if (msg:len = 2048 ) {
sl_send_reply(513, Message too big);
exit;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method==REGISTER)
record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf(P-hint: rr-enforced\r\n);
route(1);
};
if (!uri==myself) {
# mark routing logic in request
append_hf(P-hint: outbound\r\n);
route(1);
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method==REGISTER) {
save(location);
exit;
};
}
# native SIP destinations are handled using our USRLOC DB
if(method==INVITE){
if (dst_ip == 192.168.1.248)
force_rtp_proxy(oei);
if (dst_ip == 172.26.0.2)
force_rtp_proxy(oie);
t_on_reply(1);
};
if (is_method(BYE))
unforce_rtp_proxy();
if (!lookup(location,m)) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_on_failure(1);
t_reply(404, Not Found);
exit;
case -2:
sl_send_reply(405, Method Not Allowed);
exit;
}
}
route(1);
}
route[1] {
# send it out now; use stateful forwarding as it works
# reliably even for UDP2TCP
failure_route[1];
if (!t_relay()) {
sl_reply_error();
};
exit;
}
onreply_route[1]{
if (status==200){
if(dst_ip == 172.26.0.2)
force_rtp_proxy(oie);
if(dst_ip == 192.168.1.248)
force_rtp_proxy(oei);
}
}
failure_route[1]{
unforce_rtp_proxy();
}
when i make call and check on rtpproxy debug and see the rtpproxy debug :
DBUG:handle_command: received command 18781_4
UIEc0,18,4,97,9,2,15,8,101 09d614a45c92f...@172.26.0.100 172.26.0.100
2908 824bcd8bb5ba14fa;1
INFO:handle_command: new session 09d614a45c92f...@172.26.0.100, tag
824bcd8bb5ba14fa;1 requested, type strong
INFO:handle_command: new session on a port 48190 created, tag
824bcd8bb5ba14fa;1
INFO:handle_command: pre-filling caller's address with 172.26.0.100:2908
DBUG:doreply: sending reply 18781_4 48190 192.168.1.248
DBUG:handle_command: received command 18780_4 LEIc0,101
09d614a45c92f...@172.26.0.100 192.168.1.6 17206 824bcd8bb5ba14fa;1
49ee0e488eccead5;1
INFO:handle_command: lookup on ports 48190/42508, session timer restarted
INFO:handle_command: pre-filling callee's address with 192.168.1.6:17206
DBUG:doreply: sending reply 18780_4 42508 172.26.0.2
INFO:process_rtp: session timeout
INFO:remove_session: RTP stats: 238 in from callee, 323 in from
caller, 561 relayed, 0 dropped
INFO:remove_session: RTCP stats: 1 in from callee, 0 in from caller, 1
relayed, 0 dropped
INFO:remove_session: session on ports 48190/42508 is cleaned up
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
[OpenSIPS-Users] Need help Nathelper + rtpproxy
Hi all
i set up rtpproxy run in same machine with opensips
my network topology:
ip phone1 (192.168.1.6)
(192.168.1.248)opensips(172.26.0.2)---(172.26.0.100)ip phone 2
media :
ip phone1 (192.168.1.6)
(192.168.1.248)rtpproxy(172.26.0.2)---(172.26.0.100)ip phone 2
i start rtpproxy :
rtpproxy -l 172.26.0.2/192.168.1.248 -f -F -s udp:127.0.0.1:2 -d
DBUG:LOG_LOCAL7
the IP Phone 2 call IP Phone 1 and i did successfull on signaling + media
when i disconnect the call i didnt see the command tear down the media session
on rtpproxy
it is normal or i mis-config the opensips.cfg, please help
Thank you
Ha
here is my opensips.cfg:
# --- global configuration parameters
debug=9 # debug level (cmd line: -dd)
fork=yes
log_facility=LOG_LOCAL7
log_stderror=no # (cmd line: -E)
children=4
port=5060
# -- module loading --
#set module path
mpath=/usr/local/lib/opensips/modules/
loadmodule db_mysql.so
loadmodule signaling.so
loadmodule sl.so
loadmodule tm.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule usrloc.so
loadmodule registrar.so
loadmodule textops.so
loadmodule mi_fifo.so
loadmodule uri.so
loadmodule xlog.so
loadmodule nathelper.so
#loadmodule snmpstats.so
# - setting module-specific parameters ---
# -- mi_fifo params --
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
# -- usrloc params --
#modparam(usrloc, db_mode, 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam(usrloc, db_url, mysql://opensips:opensip...@localhost/opensips)
modparam(usrloc, db_mode, 2)
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam(rr, enable_full_lr, 1)
modparam(nathelper, rtpproxy_sock, udp:127.0.0.1:2)
modparam(nathelper, nortpproxy_str, )
# - request routing logic ---
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
};
if (msg:len = 2048 ) {
sl_send_reply(513, Message too big);
exit;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method==REGISTER)
record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf(P-hint: rr-enforced\r\n);
route(1);
};
if (!uri==myself) {
# mark routing logic in request
append_hf(P-hint: outbound\r\n);
route(1);
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method==REGISTER) {
save(location);
exit;
};
}
# native SIP destinations are handled using our USRLOC DB
if(method==INVITE){
if (dst_ip == 192.168.1.248)
force_rtp_proxy(oei);
if (dst_ip == 172.26.0.2)
force_rtp_proxy(oie);
t_on_reply(1);
};
if (is_method(BYE))
unforce_rtp_proxy();
if (!lookup(location,m)) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_on_failure(1);
t_reply(404, Not Found);
exit;
case -2:
sl_send_reply(405, Method Not Allowed);
exit;
}
}
route(1);
}
route[1] {
# send it out now; use stateful forwarding as it works
# reliably even for UDP2TCP
failure_route[1];
if (!t_relay()) {
sl_reply_error();
};
exit;
}
onreply_route[1]{
if (status==200){
if(dst_ip == 172.26.0.2)
force_rtp_proxy(oie);
if(dst_ip == 192.168.1.248)
force_rtp_proxy(oei);
}
}
failure_route[1]{
unforce_rtp_proxy();
}
when i make call and check on rtpproxy debug and see the rtpproxy debug :
DBUG:handle_command: received command 18781_4 UIEc0,18,4,97,9,2,15,8,101
09d614a45c92f...@172.26.0.100 172.26.0.100 2908 824bcd8bb5ba14fa;1
Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
Hi Bogdan
i just make a test to see what happen with opensips + rtpproxy
ip phone 1(192.168.1.5), ip phone 2(192.168.1.9) opensips +
rtpproxy(192.168.1.248)
i use the default opensips.cfg and edit some lines:
loadmodule nathelper.so
modparam(nathelper, rtpproxy_sock, udp:localhost:2) route {.
--- On Tue, 12/22/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, December 22, 2009, 3:50 AM
Hi Ha,
well, depends - do you want your opensips to talk to the outside world?
if no, you do not need a WAN address.
Regards,
Bogdan
ha do wrote:
Hi all
i install rtpproxy in same Opensips machine
Do i really need at least 1 Wan IP address for Opensips + rtpproxy work
Thank you
Ha`
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
Hi Bogdan
i just make a test to see what happen with opensips + rtpproxy
ip phone 1(192.168.1.9), ip phone 2(192.168.1.5), opensips +
rtpproxy(192.168.1.248)
i use the default opensips.cfg and edit some lines:
loadmodule nathelper.so
modparam(nathelper, rtpproxy_sock, udp:localhost:2)route {.
if(has_totag()){
if (is_method(BYE)){..
}else if (is_method(INVITE)){
force_rtp_proxy();
record_route();
}
}
.
}
when i make call call from IP phone 1 to IP phone 2, and media go directly from
ip phone 1 to ip phone 2
Media is not go through the rtpproxy
what should i do to force media go through the rtpproxy(just test)
1 more question on the flag:
from the alg.cfg
force_rtp_proxy(FAII), force_rtp_proxy(FAIE),
force_rtp_proxy(FAEI), force_rtp_proxy(FAEE),
i read on the nathelper module : the flag is Lower case - does it still or i
have to change to the flag to lower case
i run rtpproxy :
rtpproxy -l 192.168.1.248 -s udp:192.168.1.248:2 -d DBUG:LOG_LOCAL7 -F
[r...@localhost run]# ll
total 108
drwxr-xr-x 2 root root 4096 May 25 2008 console
drwxr-xr-x 2 root root 4096 Dec 22 17:35 dbus
-rw-r--r-- 1 root root 5 Dec 22 17:35 haldaemon.pid
-rw--- 1 root root 5 Dec 22 17:35 klogd.pid
-rw-r--r-- 1 root root 5 Dec 22 17:35 messagebus.pid
drwxr-xr-x 2 mysql mysql 4096 Dec 22 20:55 mysqld
drwxrwxr-x 2 root root 4096 Jun 15 2008 netreport
drwxr-xr-x 2 root root 4096 May 28 2008 pm
-rw-r--r-- 1 root root 5 Dec 23 17:30 rtpproxy.pid
drwxr-xr-x 2 root root 4096 Dec 18 16:35 setrans
-rw-r--r-- 1 root root 5 Dec 22 17:35 sshd.pid
-rw--- 1 root root 5 Dec 22 17:35 syslogd.pid
-rw-rw-r-- 1 root utmp 4992 Dec 23 15:56 utmp
and i get message :
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: DBG:tm:t_newtran:
transaction on entrance=0x
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:handle_command: received command
5459_4 D 5772c9303b3d7...@192.168.1.9 2adb6fd4a0d3fcb2 517a7cba18e3441a
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=
Dec 23 16:35:37 localhost rtpproxy[5126]: INFO:handle_command: delete request
failed: session 5772c9303b3d7...@192.168.1.9, tags
2adb6fd4a0d3fcb2/517a7cba18e3441a not found
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=78
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:doreply: sending reply 5459_4
E8
Thank you
Ha
--- On Tue, 12/22/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, December 22, 2009, 3:50 AM
Hi Ha,
well, depends - do you want your opensips to talk to the outside world?
if no, you do not need a WAN address.
Regards,
Bogdan
ha do wrote:
Hi all
i install rtpproxy in same Opensips machine
Do i really need at least 1 Wan IP address for Opensips + rtpproxy work
Thank you
Ha`
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
Hi Bogdan
please ignore :
and i get message :
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: DBG:tm:t_newtran:
transaction on entrance=0x
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:handle_command: received command
5459_4 D 5772c9303b3d7...@192.168.1.9 2adb6fd4a0d3fcb2 517a7cba18e3441a
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=
Dec 23 16:35:37 localhost rtpproxy[5126]: INFO:handle_command: delete request
failed: session 5772c9303b3d7...@192.168.1.9, tags
2adb6fd4a0d3fcb2/517a7cba18e3441a not found
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=78
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:doreply: sending reply 5459_4
E8
because i change :
if (loose_route()) {
if (is_method(BYE)) {
unforce_rtp_proxy();
and i still need help on media + the flag
Thank you
Ha
--- On Tue, 12/22/09, ha do haloha...@yahoo.com wrote:
From: ha do haloha...@yahoo.com
Subject: Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, December 22, 2009, 8:38 PM
Hi Bogdan
i just make a test to see what happen with opensips + rtpproxy
ip phone 1(192.168.1.9), ip phone 2(192.168.1.5), opensips +
rtpproxy(192.168.1.248)
i use the default opensips.cfg and edit some lines:
loadmodule nathelper.so
modparam(nathelper, rtpproxy_sock, udp:localhost:2)route {.
if(has_totag()){
if (is_method(BYE)){..
}else if (is_method(INVITE)){
force_rtp_proxy();
record_route();
}
}
.
}
when i make call call from IP phone 1 to IP phone 2, and media go directly from
ip phone 1 to ip phone 2
Media is not go through the rtpproxy
what should i do to force media go through the rtpproxy(just test)
1 more question on the flag:
from the alg.cfg
force_rtp_proxy(FAII), force_rtp_proxy(FAIE),
force_rtp_proxy(FAEI), force_rtp_proxy(FAEE),
i read on the nathelper module : the flag is Lower case - does it still or i
have to change to the flag to lower case
i run
rtpproxy :
rtpproxy -l 192.168.1.248 -s udp:192.168.1.248:2 -d DBUG:LOG_LOCAL7 -F
[r...@localhost run]# ll
total 108
drwxr-xr-x 2 root root 4096 May 25 2008 console
drwxr-xr-x 2 root root 4096 Dec 22 17:35 dbus
-rw-r--r-- 1 root root 5 Dec 22 17:35 haldaemon.pid
-rw--- 1 root root 5 Dec 22 17:35 klogd.pid
-rw-r--r-- 1 root root 5 Dec 22 17:35 messagebus.pid
drwxr-xr-x 2 mysql mysql 4096 Dec 22 20:55 mysqld
drwxrwxr-x 2 root root 4096 Jun 15 2008 netreport
drwxr-xr-x 2 root root 4096 May 28 2008 pm
-rw-r--r-- 1 root root 5 Dec 23 17:30 rtpproxy.pid
drwxr-xr-x 2 root root 4096 Dec 18 16:35 setrans
-rw-r--r-- 1 root root 5 Dec 22 17:35 sshd.pid
-rw--- 1 root root 5 Dec 22 17:35 syslogd.pid
-rw-rw-r-- 1 root utmp 4992 Dec 23 15:56 utmp
and i get message :
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: DBG:tm:t_newtran:
transaction on entrance=0x
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:handle_command: received command
5459_4 D 5772c9303b3d7...@192.168.1.9 2adb6fd4a0d3fcb2 517a7cba18e3441a
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=
Dec 23 16:35:37 localhost rtpproxy[5126]: INFO:handle_command: delete request
failed: session 5772c9303b3d7...@192.168.1.9, tags
2adb6fd4a0d3fcb2/517a7cba18e3441a not found
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=78
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:doreply: sending reply 5459_4
E8
Thank you
Ha
--- On Tue, 12/22/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re:
[OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, December 22, 2009, 3:50 AM
Hi Ha,
well, depends - do you want your opensips to talk to the outside world?
if no, you do not need a WAN address.
Regards,
Bogdan
ha do wrote:
Hi all
i install rtpproxy in same Opensips machine
Do i really need at least 1 Wan IP address for Opensips + rtpproxy work
Thank you
Ha`
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
-Inline Attachment Follows-
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi
Re: [OpenSIPS-Users] Help with sip trace errors please
Hi Bogdan the scripts/dbtext/opensips/sip_trace works without creating new sip_trace when opensips stop and start again :) the option modparam(db_text, db_mode, 1) doesnt help on writing :( the db_flatstore does store real time message but the INVITE, RINGING, 200ok, ACK are stored in difference files(sip_trace_9.log, sip_trace_10.log,sip_trace_11.log,sip_trace_12.log) For each table there will be several files, one file for every OpenSIPS process that wrote some data into that table Thank you very much Ha` --- On Thu, 12/17/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: From: Bogdan-Andrei Iancu bog...@voice-system.ro Subject: Re: [OpenSIPS-Users] Help with sip trace errors please To: OpenSIPS users mailling list users@lists.opensips.org Date: Thursday, December 17, 2009, 2:44 AM Hi Ha, ha do wrote: Hi Bogdan you need to take the new sip_trace file that comes with 6439 - the table format was changed, not the code in opensips. i cannt what to do, i did new complie the source and new install the opensips, i do |svn co https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.6 opensips_1_6 on new machine then do make , then do install | [bogdan] It is not about recompiling sources - you need to use the new sip_trace file with the definition of the table (from scripts/dbtext/opensips/sip_trace) | my question :|do you have a plan to make enhancement of sip_trace that logs all message without stop/start opensips how to admin opensips without log :(( example : our opensips works fine for everything, but after some days someone calls me and ask he cannot make call at 14:00PM yesterday and there is no log to check so ask him to make a call for debugging but the call is fine at debugging time if the siptrace module works without stop and start opensips, i will open the log to check what happen with him at this time :) [bogdan] probably you mean that, when using dbtext, you do not see the content of the table untill a restart of opensips, right ? if so may try using db_mode = 1 for db_text module (http://www.opensips.org/html/docs/modules/devel/db_text.html#id228234), but I'm not sure it this non-caching mode will affect the write ops also (and not only the read ops). Another solution is to use the db_flatstore driver (instead of db_text) - this one does realtime writing on file. Regards, Bogdan Thank you Ha` --- On *Tue, 12/15/09, Bogdan-Andrei Iancu /bog...@voice-system.ro/* wrote: From: Bogdan-Andrei Iancu bog...@voice-system.ro Subject: Re: [OpenSIPS-Users] Help with sip trace errors please To: OpenSIPS users mailling list users@lists.opensips.org Date: Tuesday, December 15, 2009, 12:47 AM Hi Ha, ha do wrote: Hi Bogdan i use the Checked out revision 6439. it is still issue when the sip_trace has content you need to take the new sip_trace file that comes with 6439 - the table format was changed, not the code in opensips. do you have a plan to make enhancement of sip_trace that logs all message without stop/start opensips how to admin opensips without log :(( I do not understand your question...could you rephrase ? Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] question on B2B module
Hi Anca let me describe the topology: VOIP Service Provider SIP trunk port 5070Opensips1 --sip trunk port 5060Opensips2Ip phone the Opnesips1 has 2 interface : interface 1 : IP address 192.168.1.2 -- trunking with opensips2 interface 2 : IP address 115.22.22.3 -- trunking with VOIP service provider VOIP provider IP address : 115.1.1.2 IP address of Opensips2 : 192.168.1.3 when i do top hiding if(is_method(INVITE) src_ip==115.1.1.2) $du=192.168.1.3//is it right ? b2b_init_request(top hiding); what is value i set : modparam(b2b_entities, server_address, which IP address Private or Public)Thank you Ha` --- On Wed, 12/16/09, Anca Vamanu a...@opensips.org wrote: From: Anca Vamanu a...@opensips.org Subject: Re: [OpenSIPS-Users] question on B2B module To: OpenSIPS users mailling list users@lists.opensips.org Date: Wednesday, December 16, 2009, 2:30 AM Hi Ha, Yes, such a configuration is possible. You have to set the $du to the address of OpenSIPS2 before doing b2b_init on OpenSIPS1. Regards, -- Anca Vamanu www.voice-system.ro ha do wrote: Hi all network topology: VOIP Service Provider SIP trunk port 5070Opensips1 --sip trunk port 5060Opensips2Ip phone Media : VOIP Service Provider Opensips1(rtpproxy)Ip phone can i use the B2B module on opensips1 to do topology hiding and use the rtpproxy on opensips1 to force media from IP phone to VOIP SP please advice. Thank you Ha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Help with sip trace errors please
Hi Bogdan you need to take the new sip_trace file that comes with 6439 - the table format was changed, not the code in opensips. i cannt what to do, i did new complie the source and new install the opensips, i do svn co https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.6 opensips_1_6 on new machine then do make , then do install my question :do you have a plan to make enhancement of sip_trace that logs all message without stop/start opensips how to admin opensips without log :(( example : our opensips works fine for everything, but after some days someone calls me and ask he cannot make call at 14:00PM yesterday and there is no log to check so ask him to make a call for debugging but the call is fine at debugging time if the siptrace module works without stop and start opensips, i will open the log to check what happen with him at this time :) Thank you Ha` --- On Tue, 12/15/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: From: Bogdan-Andrei Iancu bog...@voice-system.ro Subject: Re: [OpenSIPS-Users] Help with sip trace errors please To: OpenSIPS users mailling list users@lists.opensips.org Date: Tuesday, December 15, 2009, 12:47 AM Hi Ha, ha do wrote: Hi Bogdan i use the Checked out revision 6439. it is still issue when the sip_trace has content you need to take the new sip_trace file that comes with 6439 - the table format was changed, not the code in opensips. do you have a plan to make enhancement of sip_trace that logs all message without stop/start opensips how to admin opensips without log :(( I do not understand your question...could you rephrase ? Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] question on B2B module
Hi all network topology: VOIP Service Provider SIP trunk port 5070Opensips1 --sip trunk port 5060Opensips2Ip phone Media : VOIP Service Provider Opensips1(rtpproxy)Ip phone can i use the B2B module on opensips1 to do topology hiding and use the rtpproxy on opensips1 to force media from IP phone to VOIP SP please advice. Thank you Ha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Help with sip trace errors please
Hi Bogdan i use the Checked out revision 6439. it is still issue when the sip_trace has content do you have a plan to make enhancement of sip_trace that logs all message without stop/start opensips how to admin opensips without log :(( Thank you Ha` --- On Mon, 12/14/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: From: Bogdan-Andrei Iancu bog...@voice-system.ro Subject: Re: [OpenSIPS-Users] Help with sip trace errors please To: OpenSIPS users mailling list users@lists.opensips.org Date: Monday, December 14, 2009, 5:38 AM Hi Ha, found the bug - there was an issue when loading the sip_trace file with content (some columns being NULL). I fixed the problem on SVN, so if you do an update and re-compile, you should not need to delete the file during restarts. Thanks and regards, Bogdan ha do wrote: HI Bogdan it is clear now :) f the file has content (aside the first line), the siptrace module does not start ? no, the sip_trace module starts fine, i only need to replace new sip_trace file then the error is gone :) is there a module with the same function as sip_trace but no need to shutdown opensips Thank you Ha` --- On *Fri, 12/11/09, Bogdan-Andrei Iancu /bog...@voice-system.ro/* wrote: From: Bogdan-Andrei Iancu bog...@voice-system.ro Subject: Re: [OpenSIPS-Users] Help with sip trace errors please To: OpenSIPS users mailling list users@lists.opensips.org Date: Friday, December 11, 2009, 11:18 AM Hi Ha, Note that the DB content is updated only at shutdown. So, to synthesize - if the file has content (aside the first line), the siptrace module does not start ? Regards, Bogdan ha do wrote: HI Brian [r...@localhost opensipsdb]# ls -ld /tmp/opensipsdb/ drwxr-xr-x 2 root root 4096 Dec 11 14:56 /tmp/opensipsdb/ [r...@localhost opensipsdb]# ls -ld /tmp/opensipsdb/sip_trace -rw-r--r-- 1 root root 168 Dec 11 14:56 /tmp/opensipsdb/sip_trace [r...@localhost opensipsdb]# ps axu | grep opensips root 2014 0.0 1.1 38072 2928 ? S 14:56 0:00 /usr/local/sbin/opensips root 2016 0.0 0.3 38072 824 ? S 14:56 0:00 /usr/local/sbin/opensips root 2017 0.0 0.3 38072 796 ? S 14:56 0:00 /usr/local/sbin/opensips root 2018 0.0 0.3 38072 796 ? S 14:56 0:00 /usr/local/sbin/opensips root 2019 0.0 0.3 38072 796 ? S 14:56 0:00 /usr/local/sbin/opensips root 2020 0.0 0.6 38072 1712 ? S 14:56 0:00 /usr/local/sbin/opensips root 2021 0.0 0.6 38072 1696 ? S 14:56 0:00 /usr/local/sbin/opensips root 2022 0.0 0.6 38072 1724 ? S 14:56 0:00 /usr/local/sbin/opensips root 2023 0.0 0.6 38072 1588 ? S 14:56 0:00 /usr/local/sbin/opensips root 2024 0.0 0.2 38072 632 ? S 14:56 0:00 /usr/local/sbin/opensips root 2025 0.0 0.3 38072 948 ? S 14:56 0:00 /usr/local/sbin/opensips root 2026 0.0 0.3 38076 824 ? S 14:56 0:00 /usr/local/sbin/opensips root 2027 0.0 0.3 38072 892 ? R 14:56 0:00 /usr/local/sbin/opensips root 2028 0.0 0.3 38072 892 ? S 14:56 0:00 /usr/local/sbin/opensips root 2029 0.0 0.3 38072 892 ? S 14:56 0:00 /usr/local/sbin/opensips root 2030 0.0 0.3 38072 892 ? S 14:56 0:00 /usr/local/sbin/opensips root 2031 0.0 0.3 38072 880 ? S 14:56 0:00 /usr/local/sbin/opensips root 2044 0.0 0.2 3904 672 pts/0 R+ 15:00 0:00 grep opensips let me describe my steps: step1 : create file sip_trace then insert with 1 line : id(int,auto) time_stamp(int) callid(string) traced_user(string) msg(string) method(string) status(string) fromip(string) toip(string) fromtag(string) direction(string) step 2 : start opensips with sip_trace enable : /etc/init.d/opensips start step 3 : use xlite softphone(2000) + eyebeam(8000) register and register successfull step 4 : make call from 2000 to 8000 then connected fine then hangup step5 : check the /tmp/opensipsdb/sip_trace : cat /tmp/opensipsdb/sip_trace and result [r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace id(int,auto) time_stamp(int) callid(string) traced_user(string) msg(string) method(string) status(string) fromip(string) toip(string) fromtag(string) direction(string) step 6: wait about 3 mins step 7: check the sip_trace file : [r...@localhost opensipsdb]# cat /tmp/opensipsdb/sip_trace id(int,auto
